Active noise control (ANC) is one of the techniques for silencing noise. The ANC is the technology of silencing noise by interfering with noise using sound waves (control sound) with an equal amplitude and an inverse phase.
Recently, an active silencer is used to silence noise of an air-conditioner, in a factory, in a vehicle, etc.
Described below are typical conventional active silencers.
The patent document 1 discloses an active silencer having high silencing performance with low computational complexity. The active silencer is configured by a sensor microphone 101, an FIR filter 102 that can be set with a variable filter coefficient, an FIR filter 103 with a fixed filter coefficient, an LMS arithmetic unit 104 provided at the stage after the FIR filter 103, a controlling speaker 105, and an error microphone 106 as illustrated in FIG. 11. An adaptive filter 107 is configured by the FIR filter 102, the FIR filter 103, and the LMS (least mean square) arithmetic unit 104.
The sensor microphone 101 detects a signal (reference signal) corresponding to noise, and outputs the signal to the FIR filter 102 that can be set with a variable filter coefficient and the FIR filter 103 having a fixed filter coefficient.
The FIR filter (filter of an error path) 103 having a fixed coefficient holds input reference signals x(t) both at the current time and in the past for the number of its taps. The signal (filter reference signal) r(t) obtained by convoluting the propagation function w^{right arrow over ( )}=[w^(1), w^(2), . . . , w^(Nw)] of the error path from the controlling speaker 105 to the error microphone 106 to the x{right arrow over ( )}(t)=[x(t), x(t−1), . . . , x(t−Nw+1)] obtained by expressing the reference signal x(t) by vector by the following equation (1).r(t)=w^{right arrow over ( )}*x(t)  (1)
(* indicates a convolution arithmetic.)
The LMS arithmetic unit 104 holds the input reference signals r(t) input from the FIR filter 103 both at the current time and in the past for the number (Nh) of the taps of the FIR filter 102. Then, the coefficient h{right arrow over ( )}(t+1)=[h(1, t+1), h(2, t+1), . . . , h(Nh, t+1)] of the FIR filter 102 at the next time point is obtained by the following equation (2) using r{right arrow over ( )}(t)=[r(t), r(t−1), . . . , r(t−Nh+1)] obtained by expressing the filter reference signal by vector, and the coefficient h{right arrow over ( )}(t)=[h(1, t), h(2, t), . . . , h(Nh, t)] of the FIR filter 102 at the current timeh{right arrow over ( )}(t+1)=h{right arrow over ( )}(t)+μ·e(t)·r{right arrow over ( )}(t)  (2)
However, e(t) is a remaining noise signal detected by the error microphone 106 at the time t, and μ indicates a step size parameter.
As illustrated in FIG. 11, and as compared with an LMS algorithm, a Filtered-X LMS algorithm is obtained by adding the FIR filter 103 with a fixed coefficient at the stage before the LMS arithmetic unit 104 in the adaptive filter 107. The basic principle of the algorithm is to update (determine) the filter coefficient of the FIR filter 102 in the steepest descent method to decrease remaining noise by considering the transfer function from the controlling speaker 105 to the error microphone 106.
The Filtered-X LMS algorithm is described in, for example, the non-patent document 1.
Generally, in an adaptive algorithm for a time area such as the Filtered-X LMS algorithm etc., an amount of silenced noise is larger in a frequency band at a higher sound pressure level. Accordingly, there is the problem that an effective silencing effect cannot be obtained when there is disagreeable noise for humans in a frequency band at a low sound pressure level.
To solve the problem, in the patent document 2, the reference signal x from a sensor microphone 111 is divided into a plurality of bands x1, x2, . . . , xn, through a band division unit 112 as illustrated in FIG. 12, and the remaining noise signal e from an error microphone 116 is divided into a plurality of bands e1, e2, . . . , en, through a band division unit 114. In an adaptive filter unit 113 having a plurality of adaptive filters, a filter coefficient is updated (determined) for each band and a control signal to be output to a controlling speaker 115 is generated. Thus, a high silencing effect is obtained in a wide frequency band.
However, in the active silencer, a sufficient amount of silenced noise may not be acquired at some frequencies due to the aging of a controlling speaker and a microphone, the fluctuation of the spatial transmission system of an error path from a controlling speaker to an error microphone, disturbance noise mixed into the active silencer, etc.
In this case, there is a larger difference between a sound pressure level of a frequency band at which a sufficient amount of silenced noise can be obtained and a sound pressure level of a frequency band at which a sufficient amount of silenced noise can be obtained. As a result, as illustrated in FIG. 13, there can be the problem that a sound pressure level in each frequency band that is initially flat becomes partially outstanding as a non-silenced band after the sufficient lapse of time when the active silencer operates, thereby generating noisy sound with an outstanding non-silenced band.
In addition, when a filter coefficient is updated (determined) independently for each divided band (for example, in the case in FIG. 12), there occurs a more obvious problem in which it sounds exceedingly noisy in a non-silenced band.    Patent Document 1: Japanese Patent Publication No. 2872545 “Active Silencer”    Patent Document 2: Japanese Patent Publication No. 2517150 “Active Silencer”    Non-patent Document 1: B. Widrow and S. Stearns, “Adaptive Signal Processing”, Prentice-Hall, Englewood, Cliffs, N.J., 1985